Opensips trunk sip trabalhos

We are an android developer We have an android app that receives video calls from hardware ( intercom system). Our app uses linphone client to receive calls. - Linphone Android...are an android developer We have an android app that receives video calls from hardware ( intercom system). Our app uses linphone client to receive calls. - Linphone Android works perfectly over Wifi network, but when we switch to 3G/4G it stop working. No SIP registration happen or any incoming call. - Switching of network needs to work smoothly. Need to solve this problem and only expert who already solve this problem place the bid. We are an Android developer and we are looking for quick solution. DO NOT BID you're not expert on SIP, Linphone, and Android. Our team will assist you with an...

I need a tree trunk that is cut on both sides,This trunk should have a branch,Give all the details to the tree trunk,I'd like file in zbrush,no texture needed,The trunk of the tree should be dry

we need to build a sip client for android applications that sends imei as username and password

we need to build a sip client for andoid and sends the imei as username and password and connects to a single hostname

I need a softphone (ish) application that can monitor a Sip extension (freeswitch/fusionpbx) display a popup showing incoming call with caller ID number and name. When the call is answered by clicking the answer option on the popup or when answered via the IP desk phone (yealink t46g) I need it to open a url in a default web browser and have the ability to append the callerID number at the end of the url. (Api integration, and the ability to open a program would be great, but not needed at this time) When the call is answered by clicking the popup I need it to control a yealink phone (t46g) send the call to the yealink phone as an answered call on speaker phone... Click to call on the computer to be able to dial to the yealink phone is also needed. What I am trying to do use to be ...

I would like to talk to an expert on VOIP SIP TWILIO PLIVO

Hello I have an asterisk pbx that works behind firewall and the way it works is with a router that allows IP traffic destined for SIP communications from internet on dedicated WAN port to be directed with no inspection to and internal IP of the asterisk PBX, also all internal lan trafic to asterisk PBX is thru the internal IP address. In the router all outbound SIP traffic is redirected thru the same inbound WAN connection solely for SIP communication. This works fine with and Linksys RVS042 but I need to change to a mikrotik RB2011iLS-IN and what I would need are instructions on the changes to be made. No external access will be provided, only instructions and network topology can be given.

Gov. jay Inslee driving a car, running over black protester holding sign that states we want our fare share. black protester under the car. sign on the car that reads I502 white only. bags of money in the trunk, Inslee saying it works for me. marijuana leaf some where on the car.

we need a mobile app developer to create a softphone app for prepaid platform our budget is 100 dollar

We're developing an app which has an unepected need for Asterisk integration. We need 1 extra developer to help setup asterisk with SIP trunk to take data from a python program, and make a phone call with that data. The call will need SpeechRecognition implmeneted, and play DTMF tones.

Need a app developer for Shopify. App will work very similar to Shopify Apps like Trunk Realtime Stock Sync & Bundling or Stock Sync Supplier/Drop from Shopify app store. App will have similar features. I would like the app to basically sync with my store as a supplier store, and allow others to install app onto there shopify store and download my shopify products onto their store. They can choose what to download by category or individual items. Some app features will be to update stock quantity for imported items from my supplier website to user website. More details can be provided later. App does not need a payment processing. In short it will be to take items from my shopify and allow others to download them to their stores by category, or individually. And it will need to...

hello i would like to setup my sim bank with my gateway 32 port and softswitch sip i have sim bank 128 and 2 gateways 32 goip

We need to know how many SIP trunks are currently being used on a TDE100

Hi SARTTY, you completed my logo a year ago and I would like to offer you a project. You made the black Cat Logo for Superstitions Java. I need to add my slogan to your Logo: Sip Daringly, est. 2020 Reno, NV

Gelen Voip Çağrıları Cevaplayarak Ses mesajını dinleyip kayıt yapacak ve gelen mesaj karşısında cevap dinletecek. Eş zamanlı 500+ üzeri çağrıyı cevaplayabilmeli C# Dilin de winform uygulama olacak kaynak kod için teklif istiyorum

Hi All, Need a Engineer who can configure SIP Video Gateway on already installed Jitsi. Requirement: 1. SIP Clients on Public Internet should be able to do video calls Jitsi meet, share content, talk to Jiti meet participants without issues 2. Jitsi on Public Internet should be able to do video calls SIP Clients with SIP address, share contet. i know we can do this using sip gateway module on jitsi but we could not achive this. Note: Only for freelancers who have expirence in Jitsi and who has successfully configured this before as project milestone as payment will only be released if freelancer completes this requirement.

Hi All, Need a Engineer who can configure SIP Video Gateway on already installed Jitsi. Requirement: 1. SIP Clients on Public Internet should be able to do video calls Jitsi meet, share content, talk to Jiti meet participants without issues 2. Jitsi on Public Internet should be able to do video calls SIP Clients with SIP address, share contet. i know we can do this using sip gateway module on jitsi but we could not achive this. Note: Only for freelancers who have expirence in Jitsi and who has successfully configured this before as project milestone as payment will only be released if freelancer completes this requirement.

Hi Dritan R., We need support and consulting with the voipnow phone system. We want to have a contact to collaborate with us. Our company began offering hosted sip trunk and pbx services, including fax, voicemail, queue, IVR, unified communications, among others. We would like to know how you could collaborate and what your fees for your services would be.

Necesito un scriptlet que me permita llamar a un número telefónico, ponerlo en espera, llamar a otro número telefónico, unir ambas llamadas para iniciar una conferencia tripartita y después dejar en llamada a ellos, cuento con un servidor FreePBX y debe ser web, por lo tanto he buscado hacerlo con sipml5, y jssip, pero en todos solo he logrado funcionalidades parciales

...need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Whatsapp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/Whatsapp calls. Functional flow 1) Calls originating will send to Whatsapp gateway 2) Whatsapp gatewa...

Are you able to configure Zoho Desk with WebRTC clients to use an arbitrary SIP provider? You are the person for the job. You configure Zoho Desk Phone integration with arbitrary SIP provider we are using currently. The job is complete if we can make outbound calls using the Zoho Desk WebRTC client and receive inbound calls from SIP provider in Zoho Desk and process them in a example call flow and answer by ZohoDesk WebRTC client. We are currently using Twilio phone integration but we believe at time of this post Zoho does not have a way to control what trunk is used in Twilio.

We are looking for an in-house call centre solution. We will provide our own server. The objective is to have our own PBX system as well as call centre system for the customer care department. The tasks include: 1. Install GoAutoDial or Issabel in the server. 2. Connect with the telco provider (SIP link) and configure hotlines numbers for both call centre and PBX. 3. Configure IVR, ACD as per the attached IVR flow. 4. Configure voice recording for all call centre agents. 5. Configure the IP PBX as per the attached requirements. 6. Create the API so we can use the feature of "Click to Dial" from our CRM (web based application) 7. Info-popup, that will fetch the customer name from the CRM DB when a call comes in. This will allow our agents to greet the caller by name....

We are an android developer We have an android app that receives video calls from hardware ( intercom system). Our app uses linphone client to receive calls. - Linphone Android works perfectly over Wifi network, but when we switch to 3G/4G it stop working. No SIP registration happen or any incoming call. - Switching of network needs to work smoothly. Need to solve this problem and only expert who already solve this problem place the bid. We are an Android developer and we are looking for quick solution. DO NOT BID you're not expert on Linphone Android. Our team will assist you with anything you might need.

I need a programer to change the layout of a open source SIP VoIP app and publishing it to the Google play store an apple iTunes.

Hello there, I want a Sip Server or Sip Phone ready made code. Functionalities that I need the most is: 1) When one user calls another user through our server and that user does not exists server must call one exe. 2) Sip phone must have a feature to accept multiple calls on one device redirecting calls to specified list of microphones and speakers in a text file. Note : We do not want to develop it from scratch. We do not mind any technology working on this but there are two terms:- 1) Code should give us a good control and should have good code standard. 2) Code should allow us to run server and sip phone on Windows system.

I need android walkie talkie app with GPS tracking. Backend with SIP protocol, and frontend in Laravel for manage user/channel/maps/video.

I am looking to build a product like with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all the bullshit. I will provide sample layout of homepage and dashboard to build

I'm looking for somebody that can install siptrunk on my aws. The problem is that I need a static ip but i can not fixed it.

We have a new Cisco UC560 that needs configuration. Device - Cisco UC 560 Needs to configure 1 SIP trunk and 6 Extensions Phase 1 -Getting the SIP trunk up -Getting the Extensions configured -Internal calls between extensions to be working perfectly Phase 2 -Ring groups -External calls incoming and outgoing to be configured -External caller IDs to be configured -Call transfers to work fine Phase 3 -Small configuration touchups - after 7 days Extra $$$ for quick completion

I need a website built for car mats. I need my customers to be able to select their car by year, make and model and given different options for their car by color, and either 5 or 7 passenger SUV or car and or trunk.

...licence plate of the car is pre-registered in the system and an operator reads it throughout a mobile application and indicates him to drive up to the Truck area #4. Once queued in the Truck area lane waits until the Welcome Gate hostess indicates him to drive to the left or right side of the trucks reaching the Loading Area. 2 cars at once. One or two operators load one or more boxes of goods to his trunk. Once loaded, he drives towards the Exit area passing through the side of the truck queueing up in any of the 5 Exit area lanes. An operator cross-check his made purchase and loaded stuff as control until an OK signal is given. The car leaves the patio through the Exit gate. Besides each truck there are 2 operators in the Loading Area on each side of the truck. One more operato...

...caller details on your soft phone or desk phone while making calls. * Incoming and outgoing call events automatically logged inside Zoho Desk. For local soft phone we are currently using Bria 5 by Counterpath. We are also considering Zoiper 5 PRO. As supplier of managed voice services, we supply our own SIP trunks / hosted extensions that integrate perfectly with both platforms. We look forward to integrating with Zoho PhoneBridge in order to automate call logging process within Zoho CRM. Explanatory documents relating to scope of work attached. Please contact me should you be able to assist....

Hi Freelancers, We want to a Windows 10 Based SIP and XMPP client developed. You Will design and develop the software. Yes need to design a logo and icon for the product. The branding and licensing will be fully owned by ALSOFT. I will give the name later. We have two software. One is open source Linphone SIP client. I don't like the design of Linphone software. I like the Bria Professional SIP phone interface. One the other hand we use spark as a XMPP client. Spark is really a powerful XMPP client. You can use modify Spark and add the SIP functionality there. Basically I need merged version of Linphone and Spark. We use FreePBX as SIP server and Openfire as XMPP server. Regards, ALSOFT

Hello! Looking for someone who can construct for me a SIP connection system to make VOIP calls to west Europe (spain, france), Turkey and Palestine (most of the calls will be to turkey). Im not looking for something ‘large scale’, just to make a few outbound calls, with the ability to customize my CallerID freely. I have little knowledge in the technical aspect nor the servers to host it, so everything needs to be provided by your side. Contact me if you see yourself viable Oliver

Hello! Looking for someone who can construct for me a SIP connection system to make VOIP calls to west Europe (spain, france), Turkey and Palestine (most of the calls will be to turkey). Im not looking for something ‘large scale’, just to make a few outbound calls, with the ability to customize my CallerID freely. I have little knowledge in the technical aspect nor the servers to host it, so everything needs to be provided by your side. Contact me if you see yourself viable Oliver

install goautodial 4.0 in our dedicated server. configure webRTC, agents, add voip trunk.

We are implementing Jitsi sip video gateway using Jibri and Pjsip or Pjsua We need help troubleshooting: While initiating the call, Jibri is accepting the request but Pjsua/Pjsip is not starting. Need help to troubleshoot the same. We can provide error logs and necessary access.

I need you to develop some software for me. I would like this software to be developed for Windows using C or C++. The sip stack application is base on PJSIP SIP. standard

need to implement sip trunk on cisco 4321

Our product is described as a partner (chaser) for tequila, mezcal and/or beer. It consists in a medley of flavors that bring out a flavorful concentration of strong, sweet, aromatic spiciness to the drink in question. You take one sip of mezcal and then some of our product to enjoy. The color is dark red, and you can see the pepper floating around in a transparent cristal bottle. We use only natural flavors, such as: orange juice, tomato, lime, and spicy sauces. We need a label, for the exterior front of the bottle, as well as the list of ingredients on the back side, we want a modern looking, innovative, artsy, crafty, design. We need you to consider that due to recent regulations, there will be at least two big hexagons that cover part of the lable (top right). No mascots or cha...

We have a new Cisco UC560 that needs configuration. Device - Cisco UC 560 Needs to configure 1 SIP trunk and 5 - 10 Extensions Phase 1 -Getting the SIP trunk up -Getting the Extensions configured -Internal calls between extensions to be working perfectly Phase 2 -Ring groups -External calls incoming and outgoing to be configured -External caller IDs to be configured -Call transfers to work fine Phase 3 -Small configuration touchups - after 7 days

...page in a jpg and pdf format. The SIZE of each Promo Voucher will be 3 5/8 x 8.5 Use one sold background so I can cut it into three vouchers and get a bleed. I attached a sample called 'Bleed and Cut sample'. Attached are the colors of the boat - neon green and bright blue FRONT: Use the Term Boarding Pass in larger print Boarding Pass - Sample attached called Boarding Pass Attached is my SIP-n-Cycle Pedal Cruise logo. On the perforated side I need the: Promo Voucher' information To: From: Message: Promo Code Promo Value BACK: Private Parties Birthday Parties Corporate Team Building Family Reunions bachelor/bachelorette Parties BOOK NOW at 334.399.2387 Small print on the bottom: Not redeemable for cash Good until 12/31/2020 Can not be combined with

Adding some codecs to existing SIP Phone project (Java)

We are looking to customise Bitrix24 to be used for a call centre. There are 2 ways our team will use the software. 1. Each person will have a set of leads to call on a regular basis. 2. We have campaigns where leads are uploaded and each call is then handed to an agent. The pr...person will have a set of leads to call on a regular basis. 2. We have campaigns where leads are uploaded and each call is then handed to an agent. The process is outbound. We need call recording and history for each client / contact recorded on the system. There will be custom fields for data entry for each new campaign. Daily / Weekly and Monthly reporting needed. Integrate our VoIP / SIP channels to dial out from software If you have other suggestions for software then we are open to loo...

5 VLAN configuration, 2 Trunk Port and Fail over.

I need some calculator on my wordpress site....( SIP CALCULETOR, LUMPSUM CALCULETOR, FD CALCULETOR , PPF CALCULETOR, INCOME TAX CALCULETO )Only bid if you can do the job.

I have my own auto dailer astrisk but i fase issue i set call limit 3 or 4 min duration but all calls ended with 120 sec this my agi code if u know how to solve please contact me mysqli_query($conn,"UPDATE `static` SET `number`='".$calledNumber."' , ...FROM `products` WHERE `product_id`='".$row_trunk['product_id']."' LIMIT 1"); $row_products = mysqli_fetch_assoc($result_products); $agi->set_variable("trunk_id", $row_trunk['product_id']); $agi->exec('Set', "CALLERID(all)=$calledNumber"); $statusDial = $agi->exec_dial('SIP',$row_trunk[�...

We are looking for a C++ Senior Developer who is very good expertise in PJ SIP. We are porting our existing SIP applications from using the Radvision SIP stack to using PJ SIP. We are looking for experienced C++ developers that have worked with PJ SIP. The applicant must know SIP and work in-depth with PJ SIP The customer develops a soft-switch that provides, prepaid, Class4, and Class5 services. These are our SIP Proxy, SIP Registra, and the SIP Call Processing layers All the application run on RHEL/CentOS 7 and are multi-threaded Candidate must have at least 5+ years of experience. Candidate must have good communication skills, as he has to talk to US-based Clients. The candidate has no problem in Working Night Shi...

Hello I have on AWS instance a freePBX server installed, and sip trunk is twilio. I installed Zoiper on my android phone. I am willing to install other apps. If you need to login to server I prefer you use TeamViewer. :) I'd like to have somebody(YOU) to help me configure a softphone app on my cellphone to start making phone calls from it. Please reply with a timeframe this task would take you

ffmpeg supports vp9 svc () for normal conversion but not for rtp check line 236 (). ` avpriv_report_missing_feature(ctx, "VP9 scalability structure with multiple layers")` Need someone who can implement this feature